similar to: SIP with Delay

My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is

Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if

Any have trouble with acroread taking up massive cpu and memory? I exited my Firefox browser and the lil bastard was still hogging up my resources. Took up 69% of 4GB, and wouldn't let go, until a kill -9 showed'em, have to do it every time I open a pdf in firefox. Any use Xpdf or something else?

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

At 11:17 AM 29/10/2002 -0700, you wrote: >I'm doing a top on my server (ices2 and icecast2 from CVS) and ices is >using 14% of my cpu cycles - bastard! One channel is doing reencoing >and when I just play an ogg without reencoding. The ogg file was >encoded at 64kbs 44khz Is this praise, a bug report, a question, or other? If you're having a problem with the software, it

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

Hey list (again - annoying bastard I am) I've played with Firefly/* for a while and I have yet to find a way to have * send voicemail notification to Firefly. It appears possible using SIP (no clue whether Firefly supports it) in the sip.conf file, but there's no mention of anything voicemail-related in the IAX.conf file. I'm using IAX with Firefly, so that might just be the

Hi, We had modified some configuration in our cisco 800 series router. We set all the UDP packets from our servers to ip precedence 5 and also allocate 75% of bandwidth for UDP packets. However we still facing latency and low volume problem. Is it our 512k outbound bandwidth not enough to handle it? Thanks Regards, jorain ----- Original Message ----- From: jorain To: asterisk-users

Hello Chris, Chris Lattner wrote on 11/10/2005 at 6:44 p.m.: > It has been entirely too long since the last release, and we have > plenty of goodies for a very solid release. Do people find releases > useful, or should we just continue to run out of CVS? Does anyone > have any thoughts? Yeah, I think so :) Also, it would be really nice if an official cygwin build (the binary) was

Hi people I am new to this list - actually I am one of (probably) a few who are jumping ship from WBEL. I feel like a bit of a scab jumping over at this time - but we all make mistakes eh? I have installed CentOS and ran 'yum update' and installed everything it prompted me for. After a reboot, I realised I had booted into the same original kernel-2.4.21-20.EL.c0smp and not the updated

Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call

Hi guys, Thanks Matt for your prompt response about the C++ problem, its working great now! Again for my "virtual cdplayer", I am wondering if its possible to stop decoding within a large file, at the end of an exact sample? ie. I wish to play a song in the middle of a flac'd cd. Using the file decoder, I can seek to the start of the track(index 01 of the cds TOC), but I then want

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

Dear List, I like to assign names to vectors in a loop. Here is a short example: DMUs <- as.data.frame(matrix(c("b","c","d","a","e","h","i","f","g","j","k","l"),ncol =7, nrow=10)) colnames(DMUs) <- v_DMUs <-

Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg -------------- next part -------------- An

I am working on an app with a requirement that has yielded an unexpected problem for Rails routing. We have several controllers that handle regional data (one controller per data type - weather, demographics, etc). The problem is that there is one region who''s actions and output are different. We intend to create two controllers for each data type (i.e. weather_controller to handle

I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP?

Hello all, I've got an AVM Fritz!Card PCI that I'm using with Asterisk under chan_capi (0.3.5) Phones on our internal network all use g711u. I'm aware the chan_capi uses g711a by default. To reduce the need for transcoding, I decided to make everything use the same codec. First I changed all the phones (Cisco models 7905G, 7940G and ATA 186 all running SIP) to g711a, but it seemed